Tài liệu Lecture 22: Digital Transmission of Analog Data (Baseband Transmission): doc

CSN200 Introduction to Telecommunications, Winter 2000 Lecture-22 Digital Transmission
Digital Transmission of Analog Data (Baseband Transmission):

• Analog voice data must be translated into a series of binary digits before they can be transmitted in
digital transmission system. One means of doing this is called pulse amplitude modulation (PAM).
• The process of converting analog signal to digital is also known as A_to_D conversion or A/D
conversion. The circuit does the conversion known as A/D converter.

In analog modulation (continuous wave modulation), some parameter of a sinusoidal carrier wave is
varied continuously in accordance with the message signal.

In digital modulation (pulse modulation), some parameter of a pulse train is varied in accordance with
the message signal.


Pulse modulation:

Pulse modulation can be subdivided into two groups:
1. Analog pulse modulation
• A periodic pulse train is used as the carrier wave, and some parameter (amplitude,
duration, or position) of each pulse is varied in a continuous manner in accordance with
the corresponding sample value of the message signal.
• Information is transmitted basically in analog form (multiple amplitude), but the
transmission takes place at discrete times.
2. Digital Pulse modulation
• The message signal is represented in a form that is discrete in both time and amplitude.
• Information is transmitted in digital form as a sequence of coded pulses. The
transmission takes place at both discrete amplitude and discrete times.

Digital Pulse modulation or the use of coded pulses for the transmission of analog information represents
a basic application of digital communications.

Sampling:
• It is an operation that is basic to digital signal processing and digital communication.
• Through the use of sampling, an analog signal is converted to its digital form.
• To obtain either form of pulse modulation we start with sampling.

Sampling is the process of taking samples (amplitude) of an analog signal that are usually uniformly
spaced in time.
Sampling is usually described in time domain.

To get back the original analog signal we have to convert back the sampled signal to its unique analog
form. We could do it if we have taken enough samples of the original signal in the first place.
That means we have to choose the sampling rate properly.
The sampling theorem expresses the relation between the frequency of the analog signal and the
minimum rate of the sampling. Which is Sampling rate >= 2 x Frequency of the signal.
That means, if we sample the signal at a rate equal to or higher than twice the frequency of the signal we
can uniquely recover the signal back from the samples.
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CSN200 Introduction to Telecommunications, Winter 2000 Lecture-22 Digital Transmission

Figure-1: Sampling (a) Analog signal. (b) Sampled version of the signal.


The sampling theorem is also known as the Nyquist theorem of Sampling in case of a band limited signal.
And the sampling rate is also known as Nyquist rate.


Nyquist Theorem of Sampling for a band limited signal:

A band-limited signal of W Hz may be completely recovered from its samples taken at the rate of 2W
samples per second.

Analog pulse modulation:

Analog pulse modulation can be of three different types:
1. Pulse-amplitude modulation (PAM)
• The amplitude of regularly spaced pulses are varied in proportion to the corresponding sample
values of a continuous message signal.
• It is similar to natural sampling.
• The top of the PAM is flat.
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CSN200 Introduction to Telecommunications, Winter 2000 Lecture-22 Digital Transmission













Figure-2: Pulse Amplitude Modulation.

2. Pulse-duration modulation (PDM), also known as pulse-width modulation or pulse-length
modulation.
• The samples of the message signal vary the duration of the individual pulses.
3. Pulse-position modulation (PPM)
• The samples of the message signal vary the position of the individual pulses relative to its

unmodulated time of occurrence.
























Figure-3: (a) Modulating wave. (b) Pulse carrier.
(c) Pulse-duration Modulation. (d) Pulse-position Modulation.
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CSN200 Introduction to Telecommunications, Winter 2000 Lecture-22 Digital Transmission
Digital Pulse modulation:

Pulse Code Modulation is the most basic form of Digital Pulse Modulation.
Different form of digital pulse modulation are:
1. Pulse Code Modulation
2. Differential pulse-coded modulation
3. Adaptive differential pulse-coded modulation
4. Delta modulation
5. Delta-Sigma modulation.

Pulse Code Modulation (PCM):
With
Pulse Code Modulation, the amplitude of the sound wave is sampled at regular intervals, and is
translated into a binary number (encoding) as shown in figure.









Figure-4: Quantizing an analog signal.
7
6
5
4
3
2
1

0
111 =
110 =
101 =
100 =
011 =
010 =
001 =
000 =

The original signal is quantized into 8 levels of amplitudes. In this example we have used only eight levels
of amplitude for simplicity. These 8 amplitudes (= 2
3
) can be encoded by using only 3 bits)
But in practice we use 128 levels (= 2
7
) of voltages and use 7 bits to encode and 1 bit for parity, so in total
we use 8 bits to encode each voltage levels.

7
6
5
4
3
2
1
0
000 010 100 110 111 000 010 100 110 111 000 010 100
111 =
110 =

101 =
100 =
011 =
010 =
001 =
000 =









Figure-5: Quantized signal.








Figure-6: Pulse Coded Signal.
0 0 0 0 1 0 1 0 0 1 1 0 1 1 1 0 0 0 0 1 0 1 0 0 1 1 0 1 1 1 0 0 0 0 1 0 1 0 0
0 v
+12 v
-12 v

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CSN200 Introduction to Telecommunications, Winter 2000 Lecture-22 Digital Transmission
After quantizing, samples are taken at specific points to produce amplitude modulated pulses.
These pulses are then coded into binary/digital information as shown above.
Because we used 8 pulse levels, we needed only 3 binary positions (bits) to code each pulse.
If we had used 128 pulse levels, then 7 bits would be required plus one parity bit.

Figure-6 shows the pulse coded signal using RS232 voltage levels (+12 v and -12 v).

For digitizing a voice signal, 8000 samples per second are taken.
These 8000 samples are then transmitted as a serial stream of 0s and 1s.
Then 8 bits per sample times 8000 samples requires a 64,000 bps transmission rate.
(8 x 8000 = 64000 bps = 64 kbps)

Pulse Code Modulation is the most commonly used technique in the Pulse Modulation family.
For digitizing a voice signal, 8000 samples per second are taken.
These 8000 samples are then transmitted as a serial stream of 0s and 1s.

Now a days all of the trunk lines are digital. So, the analog signals are digitized in the end office by a
device called codec (coder-decoder) producing 8-bit number. Operation is very similar to how
modems function.


Quantization error:

• The difference between the original analog signal and the quantized signal is called quantizing error.

There are two ways to reduce quantizing error and improve the quality of the PAM signal.
• Increase the number of amplitude levels
• Sample more frequently (over sampling).

Use of Nyquist Theorem:
The codec makes 8000 samples per second (125 μsec/sample) because the Nyquist theorem says that this
is sufficient to capture all the information from a 4-kHz telephone channel bandwidth.
At a lower sampling rate, information would be lost; at a higher one, no extra information would be
gained.
For standard voice grade circuits, the sampling of 3300 Hz at an average of 2 samples/second would result
in a sample rate of 6600 samples per second.

PCM forms the heart of the modern telephone system. As a consequence, virtually all time intervals
within the telephone system are multiples of 125 μsec.

The T1 is based on PCM and consists of 24 voice channels multiplexed together. Each of the 24 channels,
in turn gets to insert 8 bits into the output stream. Where seven bits are data, and one is for control,
yielding 7x8000 = 56,000 bps of data.
A T1 frame consists of 24x8 =192 bits, plus one extra bit for frame synchronization, yielding 193 bits
every 125 μsec. This gives a gross data rate of 193x8000 = 1.544Mbps
By reducing the duration of the bit or increasing the bit rate we can accommodate more channels in the
original bit time (125 μsec) during transmission.
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